This application claims priority under 35 U.S.C. xc2xa7xc2xa7119 and/or 365 to 19991169 filed in Norway on Mar. 10, 1999; the entire content of which is hereby incorporated by reference.
The present invention relates to an arrangement for improving the speech quality, especially for VoIP (Voice over IP) calls, which arrangement comprises a Transceiver and Rate Adapter Unit (TRAU) in which an encoded speech signal from a Mobile Station (MS) is transcoded.
This is a proposal for how the speech quality can be enhanced for mobile VoIP calls. It is a well known problem that speech can be degraded for mobile calls when too many speech encodings/decodings are performed on the voice path.
There are mobile standards to overcome this for MS to MS calls, called Tandem Free Operation (TFO). The Japanese Personal Digital Cellular (PDC) system uses outband Mobile Application Part (MAP) signalling, called codec through. In Global System For Mobile Communication (GSM) there is an emerging standard, TS 04.53, while there is no TFO planned for Digital-Advanced Mobile Phone Service (D-AMPS).
These TFO standards were developed for MS to MS calls, and they do not address in an optimal way TFO over IP.
The solution described here has the following main benefits:
Improved speech quality for mobile VoIP calls compared to ordinary mobile calls
The impacts are local affecting only the MS-IP side of a MS-IP-MS call
The implementation uses outband standard #7 signalling (a standard in the CCITT Signalling system) and has minor impacts on the Mobile Switching Centre (MSC) and the Base Station Subsystem (BSS)
It can be used with all existing mobile voice codecs as long as they are supported by the IP network to which the gateway is connected
Avoids the use of transceivers in Base Station Subsystem
No need for speech coding and the use of Digital Signalling Processors (DSP) in the VoIP gateway
Tandem Free Operation (TFO) in GSM
In case of MS to MS calls in a mobile network without TFO, the speech signal is encoded within the first mobile station for transmission on the air interface, and transcoded within the first Tranceiver and Rate Adapter Unit (TRAU). The Pulse Code Modulation (PCM) samples are then transported within the fixed part of the network to the second TRAU using 64 kbit/s traffic links. The second TRAU encodes the speech signal a second time for the transmission on the second air interface. The two codecs of the connection are in xe2x80x9cTandem Operationxe2x80x9d.
This Tandem Operation has several disadvantages:
The extra encoding/decoding degrades the speech quality more than necessary
The links between the TRAUs need 64 kb/s while 16 or 8 kb/s would be sufficient
The unnecessary encoding/decoding within the TRAUs allocates Digital Signalling Processor (DSP) power
The European Telecommunications Standards Institute (ETSI) is working on a standard for TFO, TS 04.53. This standard defines inband signals between TRAUs so that TFO effects only the TRAUs and are therefore fully compatible with existing equipment.
In FIG. 1 there is illustrated in a schematical manner, the principle of Tandem Free Operation (TFO), and in the following there will be give a brief description of this principle.
TFO in GSM is defined as an inband signalling protocol between two peer transceivers. It
Tests the path for possible TFO
Establishes the TFO connection
Guarantees a fast fallback procedure
Supports resolution of Codec mismatch
The standard defines both TFO Frames (speech) and TFO messages. TFO Frames affects only the TRAUs.
For Half Rate Speech Coding (HR) the required bandwidth is 8 kbits/s using the Least Significant Bit (LSB) of each PCM sample and for Full Rate Speech Coding (FR) and Enhanced Full Rate Speech Coding (EFR) 16 kbits/s, using the two LSB of each PCM sample.
The speech quality is of special concern for a Mobile Switching Centre (MSC) based gateway. This is because of the number of encodings/decodings that can occur for IP based mobile calls.
For normal mobile calls we have the two following scenarios (using GSM as an example):
When IP is a part of a call leg the following can happen when GSM 06.10 is used as the IP audio codec:
Each encoding/decoding deteriorates the speech quality. For MS to MS calls we can hear the quality we get with two encodings. Adding one encoding as would be the case for an MSxe2x86x92IPxe2x86x92MS call will probably reduce the speech quality to an unacceptable level. One way to avoid this is by choosing G.711 as the IP codec, but then no reduction in bandwidth is achieved, which should be one of the main goals with using IP.
A main object of the present invention is to improve the speech quality, especially for mobile VOIP calls, by reducing the encoding/decoding to a minimum for thereby avoiding deterioration of speech quality.
Another object of the present invention is to also reduce the bandwidth, especially when using IP.
Another object of the present invention is to adapt the associated gateway (GW) so as to be in harmony with this optimisation.
The above objects are achieved in an arrangement as stated in the preamble, which according to the present invention is characterised by the features as stated in the enclosed patent claims.
In other words, the present invention suggests that the arrangement comprises means for either putting the TRAU in a transparent mode or letting the TRAU be bypassed altogether.
Further features and advantages of the present invention will appear from not only the enclosed patent claims, but also from the following description taken in conjunction with the enclosed drawings.